I have problem with my Asterisk (new implementation),
IP Phone (Yealink T19) able to do outbond call to PSTN via SIP Trunk,
able to talk two ways audio with called party,
but suddenly call disconnected after (around) 10 seconds,
this outbond call issue happen randomly,
somehow it happen but somehow call are normal.
Already talk with our SIP Trunk provider but they only give us codec preferences (alaw & ulaw), 5060 port, pitime 20, and SIP Trunk IP address.
And they said error code in their system indicate for error code 102 (call setup timeup failure).
There is no NAT and firewall in my environment.
Inbound call are normal.